The following account of the prior art relates to one of the areas of application of the present invention, acoustic feedback cancellation in a digital hearing aid. As is well-known, an oscillation due to acoustical feedback (typically from an external leakage path) and/or mechanical vibrations in the hearing aid can occur at any frequency having a loop gain larger than 1 (or 0 dB in a logarithmic expression), in other words for which the forward gain is larger than the leakage attenuation, AND at which the phase shift around the loop is an integer multiple of 360°. A schematic illustration of a hearing aid system is shown in FIG. 1a, the hearing aid system comprising an input transducer (here illustrated by a microphone) for receiving an acoustic input (e.g. a voice) from the environment, an analog-digital converter AD, a processing part K(z), a digital-analog converter DA and an output transducer (here illustrated by a speaker) for generating an acoustic output to the wearer of the hearing aid. The intentional signal path (or forward path) and components of the system are enclosed by the dashed outline. A frequency (f) dependent (‘external’, unintentional) acoustical feedback path GFB(f) from the output transducer to the input transducer is indicated.
Feedback reduction may e.g. be achieved                by reducing gain at frequencies, where the above criteria are met, or        by controlling the phase response around the loop to ensure a negative (rather than positive) feedback at frequencies, where the gain is large enough to cause oscillations, or        by shifting the frequency of the signal from the input to the output of the amplifier, so that an oscillation at a given frequency cannot easily build up, or        by adding an intentional feedback signal with a gain and phase response aimed at canceling the external leakage path.        
The present application deals with feedback reduction of the latter nature (cf. FIG. 1b, where y(n) is the digital input signal, u(n) is the digital output signal, K(z) represents the electrical signal path (also termed the forward path) of the hearing aid including an amplifier and processor of the input signal, GFB(f) represents the acoustical/mechanical feedback path and Ĝ(z) represents an electrical estimate of the acoustical feedback (feedback cancellation path).
Feedback cancellation systems are known in the art, including such systems using an adaptive filter in the feedback cancellation path. An example of a prior art system of this kind is illustrated in FIG. 1c. The components and signals of FIG. 1c are identical to those of FIG. 1b, except that the component Ĝ(z) in FIG. 1b representing an estimate of the acoustical feedback, in FIG. 1c is exemplified as an adaptive filter comprising a variable filter part Ĝ(z) and an algorithm or estimation part Algorithm (e.g. a Least Mean Squares (LMS) filter algorithm for determining the filter coefficients of the variable filter part Ĝ(z)). A digital probe signal, e.g. probe noise (cf. signal r(n) from the ‘Probe signal’ generator in FIG. 1c), may be used in hearing aid systems for improving determination of the feedback path from the speaker of a hearing aid to the microphone of the same hearing aid. In the embodiment of FIG. 1c, the probe signal r(n) is added to the digital output signal u(n) from the digital processing part K(z) and this signal u(n)+r(n) is fed to the output transducer and used as input to the variable filter part Ĝ(z) of the adaptive filter. The algorithm or estimation part receives as inputs to the estimation of the adaptive filter the probe signal r(n) and the digital input signal (also termed ε(n) (error signal) in the figure) to the amplifier/processing block K(z). This is known as the indirect identification method.
FIG. 2 shows a more general arrangement of the signal paths of a hearing aid system comprising feedback cancellation, where both indirect identification schemes (kr=1, ku=0, using a probe signal in the digital output) and direct identification schemes (kr=0, ku=1, no probe signal in the digital output signal used) are indicated. Alternatively, kr=ku=½, representing a scheme where equal amounts of the probe signal r(n) and the digital output signal u(n) are used as an input the algorithm part LMS. Other intermediate variants may be implemented by the arrangement in FIG. 2 (by allowing each of ku and kr to vary between 0 and 1). System identification using the indirect (kr=1, ku=0) and direct methods (kr=0, ku=1) are common knowledge in system identification and e.g. described in U. Forssell, L. Ljung, Closed-loop Identification Revisited—Updated Version, Linköping University, Sweden, LiTH-ISY-R-2021, 1 Apr. 1998.
In indirect identification, the probe signal is preferably inaudible to the user of the hearing aid. A feedback part of the probe signal is received at the microphone of the hearing aid together with ambient sound and feedback of the processed ambient sound. Hence the received signal by the microphone will be a mix of the ambient (and desired) signal and the (undesired) feedback signal from the output (including the probe noise).
The quality of the estimate of the feedback path depends on the ratio between the level of the probe signal and the level of the other signal of the microphone. The part of the microphone signal that does not originate from the probe noise will disturb the adaptation of the adaptive filter and will be called the “disturbing signal” below. The lower level of the disturbing signal, the better (more accurate) estimate or faster adaptation can be achieved.
U.S. Pat. No. 5,680,467 describes a hearing aid with an acoustic feedback compensation circuit comprising a noise generator for the insertion of noise, and an adjustable digital filter for the adaptation of the feedback signal, the adaptation involving statistical evaluation of the filter coefficients.
U.S. Pat. No. 7,013,015 describes a system for reducing feedback-conditioned oscillations in a hearing aid device, wherein microphone signals of a first microphone and of a distanced, second microphone are compared to one another. When oscillations are detected at the same frequency in both microphone signals, these oscillations are determined to be useful (non-feedback) tonal signals. Oscillations that are only present in one of the microphone signals, in contrast, are feedback-conditioned and are suppressed using suitable measures.
U.S. Pat. No. 6,549,633 describes a binaural hearing aid with signal processors in each unit, wherein a residual feedback signal representing the difference feedback signals from the actual and simulated sound processing channels is supplied and used to distinguish between howl and information sound signals of a similar character.
WO 2007/098808 describes a hearing aid with multiple microphones, directional processing means to form a spatial signal from the microphone signals, estimating means which are used to estimate feedback signals to each of the microphones and processing means to—based on the directional and feedback information—apply a gain not exceeding a resulting maximum gain limit to form a hearing loss compensation signal.